Video communications including multi-party videoconferences have become increasingly popular due to widespread deployment of broadband networks, advancements in video compression technologies, and increased availability of low-cost tools for implementing web-based video communications. One approach to implementing multi-party videoconferences with server computers is based on the web real-time communication (WebRTC) standard. This approach employs selective forwarding units (SFU(s)), which operate to forward (or route) video packets in video streams to multiple participant devices without media processing. In a typical multi-party videoconference implemented with a central server computer employing an SFU, each participant device sends at least one video stream including real-time transport protocol (RTP)/user datagram protocol (UDP) video packets over a network (e.g., the Internet) to the central server computer, which selectively forwards or routes the video streams over the network to the respective participant devices such that each participant device receives one or more video streams from one or more of the other participant devices in the multi-party videoconference.
Multi-party videoconferences that involve sending/receiving video streams including RTP/UDP video packets over networks such as the Internet are often adversely affected by changes in available bandwidths between the central server computer and the respective participant devices. For example, such changes in available bandwidths may entail a sudden drop in available egress bandwidth from the central server computer to a respective video receiver device. In such a situation, the central server computer typically detects the drop in available egress bandwidth to the respective receiver device, and sends a request to at least one of the participant devices currently sending video streams received by the respective receiver device to reduce its transmission bitrate. However, the detection of the drop in available egress bandwidth and/or the subsequent reduction of the transmission bitrate at a respective video sender device may be delayed, causing packet losses and/or packet delivery delays that may result in a suboptimal quality of experience (QoE) at the respective receiver device.